[..] |
ASTERISK-24000: chan_pjsip: Add accountcode setting |
ASTERISK-24001: res_rtp_asterisk fails to load module due to undefined symbol 'dtls_perform_handshake' when PJPROJECT is not installed |
ASTERISK-24002: No audio after WebRTC callee resumes call from hold |
ASTERISK-24003: testsuite: Write a rest_api/bridges test that puts two channels into a bridge and verifies that accountcodes/linkedids propagate |
ASTERISK-24004: HASHKEYS() doesn't work with hashes set as globals |
ASTERISK-24006: app_waitforsilence logic error when used as WaitForNoise |
ASTERISK-24007: Crash in dtls_srtp_check_pending |
ASTERISK-24008: chan_dahdi: Outgoing originated call causes crash when redirecting information is free'd after failed ast_request |
ASTERISK-24009: Disable T.38 (t38pt_udptl=no) reject T.38 REINVITE with 488 Not acceptable here |
ASTERISK-24010: Testsuite: Create a general pluggable module that performs energy detection on a specified file |
ASTERISK-24011: [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM |
ASTERISK-24015: app_transfer fails with PJSIP channels |
ASTERISK-24016: CDR logs multiple unanswered devices despite unanswered=no |
ASTERISK-24019: When a Music On Hold stream starts it restarts at beginning of file. |
ASTERISK-24020: Bridge created from AMI "Bridge" command not being destroyed when all channels leave |
ASTERISK-24021: Reference debugging has incorrect path in res_musiconhold.c |
ASTERISK-24025: intercept dtmf on meetme |
ASTERISK-24026: Syntax for GotoIf incorrectly pushed into Asterisk Wiki |
ASTERISK-24027: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up |
ASTERISK-24028: testsuite: Write a test that verifies the MixMonitor AMI action |
ASTERISK-24032: Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined |
ASTERISK-24033: [patch]Extend STAT to check relative paths (for audio files) |
ASTERISK-24034: Asterisk restarted with seg fault |
ASTERISK-24035: Asterisk Seems to Just Drop Call for no reason |
ASTERISK-24036: ARI: Recording resource should allow copying a recording |
ASTERISK-24037: ARI: RecordingFinished event should return duration of recording |
ASTERISK-24038: device state: Report ONHOLD device state if channel driver defers device state calculation to core |
ASTERISK-24039: Voicemail calls failed after 200 concurrent calls |
ASTERISK-24040: [patch]Retrieving source port of sip message in dialplan |
ASTERISK-24041: deadlock when cdr_pgsql fails to post CDR due to PGSQL connection failure |
ASTERISK-24042: Drastic LA increase while serving more than 1400 simultaneous calls |
ASTERISK-24043: ARI /continue fails to actually continue into the dialplan |
ASTERISK-24044: Unistim channel calls being dropped |
ASTERISK-24045: [patch]Voicemail to email at multiple email addresses |
ASTERISK-24046: PJSIP: No matching endpoint found on incoming call? |
ASTERISK-24047: Asterisk and RFC3264 |
ASTERISK-24048: [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts |
ASTERISK-24049: Asterisk Manager Interface: A number of list type responses aren't using astman_send_listack |
ASTERISK-24050: [patch] Improve AstDb I/O When Updating Rows |
ASTERISK-24051: [patch] Add Options To Play Beep At Start Or End Of MixMonitor |
ASTERISK-24052: app_voicemail reloads result in leaked IMAP sockets. |
ASTERISK-24053: CLONE - Problem inserting CEL records when certain characters are used |
ASTERISK-24054: CLONE - Problem inserting CEL records when certain characters are used |
ASTERISK-24058: res_fax: Manager actions do not unregister at unload |
ASTERISK-24063: [patch]Asterisk does not respect outbound proxy when sending qualify requests |
ASTERISK-24064: media formats: Improve performance in the Asterisk core by overhauling its media format architecture |
ASTERISK-24065: ooh323 irregularities in the sequence of TCS/MSD/OLC after TCS=0 |
ASTERISK-24066: res_smdi: convert to astobj2 |
ASTERISK-24067: chan_sip: upgrade registry and mwi object to ao2 |
ASTERISK-24068: Move main/manager_*.c to loadable modules |
ASTERISK-24069: Deprecate astobj.h |
ASTERISK-24070: testsuite: Fix iax2/acl_call test failure |
ASTERISK-24071: testsuite: Fix tests/channels/SIP/SDP_attribute_passthrough |
ASTERISK-24072: testsuite: Fix tests/callparking |
ASTERISK-24073: testsuite: Fix tests/feature_blonde_transfer |
ASTERISK-24074: testsuite: Fix tests/page_baseline |
ASTERISK-24075: testsuite: Fix tests/feature_attended_transfer |
ASTERISK-24082: Ensure AMI connections are torn down |
ASTERISK-24083: testsuite: Fix tests/apps/page/page_predial |
ASTERISK-24085: Documentation - We should remove or further document the 'contact' section in pjsip.conf |
ASTERISK-24087: [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy |
ASTERISK-24088: Deadlock with attended transfer of queue call to the same or other queue |
ASTERISK-24091: Testsuite - SIPpTestCase modules report success on reactor timeout. |
ASTERISK-24092: testsuite: Add channel/bridge playback control tests |
ASTERISK-24094: testsuite: Add mute/unmute tests |
ASTERISK-24096: testsuite: Add endpoint inspection tests |
ASTERISK-24097: Documentation - CHANNEL function help text missing 'linkedid' argument |
ASTERISK-24104: Certified Asterisk shouldn't have extended modules selected for build by default |
ASTERISK-24106: WebSockets Automatically decides what driver it will use |
ASTERISK-24107: [crash] chan_iax2: When building peer with allow=all |
ASTERISK-24110: r-uri not being set based on the contact from 200ok |
ASTERISK-24111: res_ari_mailboxes: Asterisk attempts to load module without res_stasis_mailbox, resulting in undefined symbol: stasis_app_mailbox_to_json |
ASTERISK-24113: Can't input Chinese in Asterisk console (context name contains Chinese characters) |
ASTERISK-24115: PJSIP: Add nominal device state tests for inter-Asterisk PUBLISH |
ASTERISK-24116: PJSIP: Add nominal MWI tests for inter-Asterisk PUBLISH |
ASTERISK-24119: HEP: Add module that exports RTCP information to a Homer Capture Server |
ASTERISK-24121: [patch] pass-through support for AMR and AMR-WB |
ASTERISK-24122: Documentaton for res_pjsip option use_avpf needs to be fixed |
ASTERISK-24123: A second instance of 'module show' from the Asterisk CLI locks Asterisk |
ASTERISK-24124: manager: UserEvent action skips over the first header |
ASTERISK-24126: testsuite: run-local creates broken symlinks |
ASTERISK-24127: chan_sip option rtpkeepalive results in comfort noise packets being sent despite flowing RTP - resulting in audible interruptions to audio |
ASTERISK-24128: [Patch] Adding default dtls settings |
ASTERISK-24131: DTLS Crash Out of libssl |
ASTERISK-24132: Remote TLS server close connection. Dialstatus = CANCEL |
ASTERISK-24133: [patch]Please support Clang; Allow no-exec stacks |
ASTERISK-24134: ARI: GET /channels/{channel_id}/variable for channel in dialplan returns 409 conflict |
ASTERISK-24136: Security: Crash in Asterisk's PJSIP code when subscribing to an event with an unexpected body type |
ASTERISK-24138: dial: Call forwarding information presented through AMI/ARI is wrong |
ASTERISK-24140: Documentation - Document IPv6 support for Asterisk as a whole. |
ASTERISK-24142: CCSS: crash during shutdown due to device lookup in destroyed container |
ASTERISK-24143: pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK |
ASTERISK-24144: Asterisk 12.4.0 Crashes with DTLS |
ASTERISK-24145: Asterisk 12.4.0 crashes randomly (Segment Fault) |
ASTERISK-24146: [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec |
ASTERISK-24147: ARI: channel hangup crashes asterisk process |
ASTERISK-24148: Can not use mysql at extensions.lua |
ASTERISK-24149: Routing problems on firewall with chan_pjsip packets on port 5060 (chan_sip and/or other port working) |
ASTERISK-24150: FRACK when using iax.conf bandwidth option. |
ASTERISK-24151: build_tools/get_documentation only picks up the first documentation block |
ASTERISK-24152: memory leak and segfault with in libpthread-2.17.so |
ASTERISK-24154: [patch]PTHREAD_MUTEX_INITIALIZER should not be assumed to be recursive mutex |
ASTERISK-24155: [patch]Non-portable and non-reliable recursion detection in ast_malloc |
ASTERISK-24156: Asterisk 'make full' fails with raise xml.dom.NotFoundErr |
ASTERISK-24158: asterisk can't validate wildcard certificate |
ASTERISK-24161: PJSIPShowEndpoint gives inaccurate count of list items |
ASTERISK-24163: Crash - in ast_hangup in channel.c - bad magic number spammed in log preceding crash |
ASTERISK-24164: High consume CPU when establishing Conf_bridge |
ASTERISK-24165: [patch]Asterisk crashes if DAHDI mfcr2 channel is destroyed while in a call |
ASTERISK-24166: SDP Asterisk. |
ASTERISK-24170: [patch] Add a manpage for the smsq application |
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility |
ASTERISK-24172: File formats/msgsm.h has no license |
ASTERISK-24173: File menuselect/menuselect_gtk.c has no license header |
ASTERISK-24174: Wiki Documentation - Features configured in features.conf - update and organize Features section |
ASTERISK-24178: [patch]fromdomainport used even if not set |
ASTERISK-24179: res_parking option parkedplay=parked results in module load failure. Values 'caller' and 'both' work fine. |
ASTERISK-24181: RLS: Large lists don't get sent because they exceed the PJSIP message length limit |
ASTERISK-24190: IMAP voicemail causes segfault |
ASTERISK-24191: Check debian patch repo for unfixed issues |
ASTERISK-24192: Multihomed asterisk b2b calls on different networks |
ASTERISK-24194: Loading AST_MODFLAG_DEFAULT in pbx_lua.c causes undefined symbol error |
ASTERISK-24195: bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge |
ASTERISK-24197: Signed integer overflow in string hash functions |
ASTERISK-24198: Typo's |
ASTERISK-24199: 'ALL' is specified in pjsip.conf.sample for TLS cipher but it is not valid |
ASTERISK-24202: rev. 420654 tcptls.c doesn't work anymore |
ASTERISK-24203: Sorcery: Allow for tests to temporarily select a sorcery wizard |
ASTERISK-24205: DTLS-SRTP fails on SIP over WebSockets call from SIPML5(chrome) to Asterisk |
ASTERISK-24208: Channels with CDR Information Remain Active Even After ConfBrige Is Ended |
ASTERISK-24209: Webrtc ICE negotiation - RTCP port is even number |
ASTERISK-24210: T.38 UDPTL Frames are sent to wrong destination IP |
ASTERISK-24211: testsuite: Fix the dial_LS_options test |
ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP engine |
ASTERISK-24215: testsuite: ARI Live Dangerously test fails due to wrong response code from Asterisk |
ASTERISK-24217: testsuite: pjsip/ami/show_subscriptions test fails |
ASTERISK-24219: testsuite: PJSIP blind transfer tests fail |
ASTERISK-24221: chan_sip: SRV lookup is not performed when using a realtime peer |
ASTERISK-24222: PJSIP: Failed assertions when placing a call with no allow= specified |
ASTERISK-24223: Gibberish Call-ID on Local channel on origination |
ASTERISK-24224: When using Bridge() dialplan application, surrogate channel appears in list and call count is inflated. |
ASTERISK-24225: Dial option z is broken |
ASTERISK-24229: ARI: playback of sounds implicitly answers channel, preventing early media playback |
ASTERISK-24231: crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable |
ASTERISK-24234: app_meetme: Crash on conference shutdown due to NULL channel passed to meetme_stasis_generate_msg() |
ASTERISK-24236: res_hep_rtcp: Module incorrectly depends on pjsip |
ASTERISK-24237: CDR: FRACK With PJSIP blonde transfer. |
ASTERISK-24238: Double free or corruption (!prev) - moh_files_generator |
ASTERISK-24239: UDP sockets used for RTP are sometimes leaked |
ASTERISK-24240: Asterisk disconnects AMI client intermittently (TCP FIN, PSH, ACK) after Action: Originate |
ASTERISK-24241: crash: CDRs recursively attempt to update Party B information in a multi-party bridge, overrunning the stack |
ASTERISK-24242: PJSIP: Error parsing/validating SDP body: Unknown error 220030 |
ASTERISK-24243: testsuite: dialog_info_xml fails on Asterisk 12 only |
ASTERISK-24245: gcc 4.1.2 complains of files that do not end with newlines |
ASTERISK-24246: Quiet warning about type qualifiers ignored on function return type |
ASTERISK-24247: Make Stasis bridges not require channels to be made departable |
ASTERISK-24249: SIP debugs do not stop |
ASTERISK-24250: [patch] Voicemail with multi-recipients To: header fix |
ASTERISK-24251: chan_sip REGISTER sent to wrong port using SRV records |
ASTERISK-24253: Attended transfers with directmedia enabled sometimes set wrong rtp address |
ASTERISK-24254: CDRs: Application/args/dialplan CEP updated during dial operation |
ASTERISK-24257: agent must dial acceptdtmf twice to bridge to queue caller |
ASTERISK-24258: Segmentation fault in ast_variable_update when using app_voicemail. |
ASTERISK-24261: Duplicate digits received from analog device |
ASTERISK-24262: AMI CoreShowChannel missing several output fields and event documentation |
ASTERISK-24263: segfault exiting when caller leaves MeetMe conference |
ASTERISK-24264: ARI: Adding a channel to a holding bridge automatically starts MOH |
ASTERISK-24265: segfault in asterisk when try to make call to IAX |
ASTERISK-24266: imapparentfolder Ignored - saving messages to another folder fails whether it exists or not |
ASTERISK-24267: Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel |
ASTERISK-24268: ${QUEUE_MEMBER(uut_queue,free)} still = 0 when agent logged in |
ASTERISK-24269: Delay in connection of speech with pjsip and webrtc |
ASTERISK-24270: Core dump while making webrtc call via pjsip |
ASTERISK-24271: Unable to make WebRTC call through chan_PJSIP nor chan_SIP |
ASTERISK-24273: ARI deviceStates inconsistency |
ASTERISK-24274: [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used |
ASTERISK-24275: Tie inter-Asterisk SIP PUBLISHes for MWI into res_mwi_external |
ASTERISK-24276: [Patch] Option to make app MOH override channel musicclass |
ASTERISK-24277: answered call logged as unanswered |
ASTERISK-24279: Documentation: Clarify the behaviour of the CDR property 'unanswered' |
ASTERISK-24280: Add 'rtpbindaddr' setting for chan_sip |
ASTERISK-24281: When bridging 2 chan_sip channels, MOH not removed from on-hold channels and bridge is never destroyed after hangup. |
ASTERISK-24282: chan_sip communicating via TCP on IP assigned to physical interface when Asterisk is bound to virtual interface/alias |
ASTERISK-24283: [patch]Microseconds precision in the eventtime column in the cel_odbc module |
ASTERISK-24284: ARI fails to strip whitespace properly on bridge type attribute, allowing for bridges to be created even when provided attribute types conflict |
ASTERISK-24285: ulimit -n 1000000,but only create 500 channels in one conference,why? |
ASTERISK-24286: Race condition in PBX core can cause missed device state changes |
ASTERISK-24287: Race conditons and other problems in res_config_pgsql |
ASTERISK-24288: [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup |
ASTERISK-24290: Endpoint identifier match value fails to parse when CIDR network format is specified |
ASTERISK-24291: res_srtp module stops working after about 35.000 processed calls |
ASTERISK-24292: Asterisk testsuite - Files that should (or not) be executable |
ASTERISK-24293: Asterisk adds ';2' to channel id in ARI events |
ASTERISK-24294: [patch] Certificate Generation Script |
ASTERISK-24295: crash: creating out of dialog OPTIONS request crashes |
ASTERISK-24296: events.json has deprecated attributes |
ASTERISK-24297: cdr.c: Minor code optimizations. |
ASTERISK-24298: CLONE - option allowexternaldomains behavior changed between release versions |
ASTERISK-24299: Segmentation fault on ast_stopstream |
ASTERISK-24300: API docs don't conform to stated Swagger version |
ASTERISK-24301: Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk |
ASTERISK-24302: Crash with PJSIP, WebRTC configuration in pjproject - pjsip_fromto_hdr_clone from /usr/local/lib/libpjsip.so |
ASTERISK-24303: asterisk crashing randomly |
ASTERISK-24304: asterisk crashing randomly because of unistim channel |
ASTERISK-24305: Crash with astobj2.c: user_data is NULL |
ASTERISK-24306: no direct media if srtp is used and thus higher cost for hosted PBX solution |
ASTERISK-24307: Unintentional memory retention in stringfields |
ASTERISK-24308: Call IDs in Asterisk 12: Planning |
ASTERISK-24310: CLONE - Wrap up time is ignored for queue members who are members in multiple queues |
ASTERISK-24311: Populating database via Alembic fails when using same database for multiple schema sets |
ASTERISK-24312: SIGABRT when improperly configured realtime pjsip |
ASTERISK-24313: Asterisk is not supporting "a=extmap" SDP attribute |
ASTERISK-24314: ConfBridge Doesn't Deliver 48 kHz Audio with Local channels |
ASTERISK-24315: [patch] pjproject: Don't compile in libyuv if --disable-video is passed to configure |
ASTERISK-24316: For httpd server, need option to define server name for security purposes |
ASTERISK-24317: Crash preceded by potential memory leak. |
ASTERISK-24319: Verify accuracy of wiki documentation on how Asterisk searches for sounds |
ASTERISK-24320: Incoming audio not working |
ASTERISK-24321: SIP deadlock when running automated queues tests |
ASTERISK-24322: Asterisk crashes on execution when compiled with MALLOC_DEBUG compiler flag - in region_data_wipe at astmm.c |
ASTERISK-24323: Bug in documentation AGI STREAM FILE CONTROL |
ASTERISK-24324: Issue compiling Asterisk13 without PJSIP |
ASTERISK-24325: res_calendar_ews: cannot be used with neon 0.30 |
ASTERISK-24326: res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted |
ASTERISK-24327: bridge_native_rtp: Smart bridge operation to softmix sometimes fails to properly re-INVITE remotely bridged participants |
ASTERISK-24328: Use of MixMonitor 'm' option results in 0 duration vm description file |
ASTERISK-24329: Music On Hold announcement cuts intro of music the first time it is played |
ASTERISK-24330: Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 |
ASTERISK-24331: Unexpected Errors in Asterisk Manager Interface Output |
ASTERISK-24332: [patch] Add ability to set "fromstring" per mailbox |
ASTERISK-24333: Crash in DTLS |
ASTERISK-24334: Crash with chan_sip on SIP to SIP over WebSockets call (WebRTC, SIPML5) |
ASTERISK-24335: [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls |
ASTERISK-24336: PJSIP timer_min_se value under 90 causes crash |
ASTERISK-24337: Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped' |
ASTERISK-24338: ARI live recording operations cause 500 Internal Server error during initial "beep" |
ASTERISK-24339: Swagger API Docs have incorrect basePath |
ASTERISK-24341: PJSIP Ability to get info per contact |
ASTERISK-24342: PJSIP: Qualifying endpoints attempts to do them all at the same time. |
ASTERISK-24343: res_corosync segfaults in dispatch_thread_handler |
ASTERISK-24344: CDR_PROP(disable) disables CDR only for first dialed party |
ASTERISK-24345: StopMixMonitor doesn't stop audio recording when run using DYNAMIC_FEATURES |
ASTERISK-24346: Using "r" option on Dial() command prevents Remote-party-ID from reaching caller |
ASTERISK-24347: can not create video conference |
ASTERISK-24348: Built-in editline tab complete segfault with MALLOC_DEBUG |
ASTERISK-24349: strictrtp has trouble in NAT scenario's with 100->183->180->200 |
ASTERISK-24350: PJSIP shows commands prints unneeded headers |
ASTERISK-24351: [patch] Allow passing options and command to MixMonitor when recording in ConfBridge |
ASTERISK-24352: Problem with SIP privacy - fromdomain |
ASTERISK-24353: DAHDI Interrupts IRQ |
ASTERISK-24354: AMI sendMessage closes AMI connection on error |
ASTERISK-24355: [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser' |
ASTERISK-24356: PJSIP: Directed pickup causes deadlock |
ASTERISK-24357: [fax] Out of bounds error in update_modem_bits |
ASTERISK-24358: chan_vpb.cc does not compile with -Werror using gcc 4.9 |
ASTERISK-24359: Deadlock in chan_sip.c when monitoring |
ASTERISK-24360: Asterisk deadlock |
ASTERISK-24361: FFA for Asterisk 13 |
ASTERISK-24362: res_hep leaks reference to configuration |
ASTERISK-24363: [patch] Add ability for Channel Drivers to provide Presence State information |
ASTERISK-24364: PJSIP realtime configuration via res_odbc with Oracle backend - Binding variable to :0 is not allowed in oracle |
ASTERISK-24365: [Patch] Dialplan function to get first/head caller channel on queue |
ASTERISK-24366: Qualify frequency on AoRs is not reflected properly on permanent contacts. |
ASTERISK-24367: PJSIP: allow all results in failure to send INVITE |
ASTERISK-24368: res_pjsip_pubsub: Subscription persistence causes crash when re-constructing stored subscription |
ASTERISK-24369: res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations |
ASTERISK-24370: res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd |
ASTERISK-24371: res_pjsip: DNS resolution on pulls SRV records for UDP transport even when DNS has entries for other transport types |
ASTERISK-24372: [patch] Add config option to play a prompt to the "winner" in app_followme |
ASTERISK-24373: Sub-second silence |
ASTERISK-24374: "sip qualify peer" CLI command stops periodic pokes for the peer forever, if the peer is unreachable |
ASTERISK-24375: pjsip reporting blank line for an expected response to a challenge in AMI event |
ASTERISK-24376: res_pjsip_refer: REFER request for remote session attempts to direct channel to external_replaces extension instead of context, without providing for the Referred-To SIP URI |
ASTERISK-24377: Dialplan pattern matching does not work as expected |
ASTERISK-24378: Release AMI connections on shutdown |
ASTERISK-24379: Testsuite: Process REF_DEBUG logs, fail any test that leaks |
ASTERISK-24380: core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs |
ASTERISK-24381: res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections |
ASTERISK-24382: chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK |
ASTERISK-24383: res_rtp_asterisk: Crash if no candidates received for component |
ASTERISK-24384: chan_motif: format capabilities leak on module load error |
ASTERISK-24385: chan_sip: process_sdp leaks on an error path |
ASTERISK-24386: Asterisk "doc/lang/language-criteria.txt" needs update or removal. |
ASTERISK-24387: res_pjsip: rport sent from UAS MUST include the port that the UAC sent the request on |
ASTERISK-24388: res_pjsip_sdp_rtp: Declining media streams should remove all lines but the media session (m) and connection line (c) |
ASTERISK-24389: chan_iax2: Unit test on Bamboo failing |
ASTERISK-24390: astobj2: REF_DEBUG reports false leaks with ao2_callback with OBJ_MULTIPLE |
ASTERISK-24391: res_fax: fax handler module reference leak |
ASTERISK-24392: res_fax: fax gateway sessions leak |
ASTERISK-24393: rtptimeout=0 doesn't disable rtptimeout |
ASTERISK-24394: CDR: FRACK with PJSIP directed pickup. |
ASTERISK-24395: DTLS Handshake between Firefox Version > 34 and Asterisk is not completed anymore |
ASTERISK-24396: Asterisk restarts after giving message 'Segmentation fault' |
ASTERISK-24397: Audiohooks require constant media flow for whispering |
ASTERISK-24398: Initialize auth_rejection_permanent on client state to the configuration parameter value |
ASTERISK-24399: Crashes - res_rtp_asterisk with DTLS-SRTP - in libssl |
ASTERISK-24400: ooh323 sends wrong hangup code |
ASTERISK-24401: Segfault in res_config_curls |
ASTERISK-24402: ARI Playbacks which use URIs that string together sounds cannot be stopped |
ASTERISK-24403: Crashed in strncasecmp from /lib/libc.so.6 - in __get_header at chan_sip.c |
ASTERISK-24404: When controlling inbound calls via AMI + async AGI, sometimes asterisk itself issues an EXEC Congestion command |
ASTERISK-24405: pjsip: assert after restart with tls |
ASTERISK-24406: Some caller ID strings are parsed differently since 11.13.0 |
ASTERISK-24408: Issue with MOH on transfers introduced in version 1.8.27 |
ASTERISK-24409: AMI action Originate not working when in sync mode |
ASTERISK-24410: res_pjsip_session: Deferred SDP Answer received in ACK is not actually processed |
ASTERISK-24411: [patch] Status of outbound registration is not changed upon unregistering. |
ASTERISK-24412: [patch]Incomplete channel originate/continue handling with ARI |
ASTERISK-24413: parking/parking_tests: Crash due to assertion in unit tests when MoH is started on channel in holding bridge |
ASTERISK-24414: TCP connection state is processed before the SIP response hits the transaction layer |
ASTERISK-24415: Missing AMI VarSet events when channels inherit variables. |
ASTERISK-24416: config option reference list in pjsip.conf.sample has missing characters |
ASTERISK-24417: New to Asterisk. Help Required to understand Call Status |
ASTERISK-24418: Chanspy Attaching to Wrong DAHDI Channel |
ASTERISK-24419: Incorrect syntax for setting language in configs/extensions.conf.sample |
ASTERISK-24420: segfault when Zoiper joins ConfBridge |
ASTERISK-24421: [ARI] appArgs not passed to both legs of local channel |
ASTERISK-24422: Monitor still alive after channel is hanged up |
ASTERISK-24423: High pitched constant sound in ConfBridge when sample rate is set to 16000 and with dsp_drop_silence=yes |
ASTERISK-24424: Warnings when ulaw/SILK16 is being transcoded |
ASTERISK-24425: [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) |
ASTERISK-24426: CDR Batch mode: size used as time value after first expire |
ASTERISK-24427: Documentation is missing for a few AMI Events - Including CDR and events triggered after the QueueStatus action |
ASTERISK-24428: Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used |
ASTERISK-24429: missing information about changes in AMI events |
ASTERISK-24430: missing letter "p" in word response in OriginateResponse event documentation |
ASTERISK-24431: Inconsistent in "QueueMember" event of "QueueStatus" action |
ASTERISK-24432: Install refcounter.py when REF_DEBUG is enabled |
ASTERISK-24433: Diversion header is only partially parsed into REDIRECTING structure |
ASTERISK-24434: Fix differing usage of assignment operators in modules.conf |
ASTERISK-24435: Asterisk 13 with TC400P segfault |
ASTERISK-24436: Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 |
ASTERISK-24437: Review implementation of ast_bridge_impart for leaks and document proper usage |
ASTERISK-24438: res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid |
ASTERISK-24439: Wrong billsec of a new call on an already answered channel |
ASTERISK-24440: Call leak in Confbridge |
ASTERISK-24441: Unable to store voicemail greetings using PostgreSQL with ODBC |
ASTERISK-24442: Outgoing call files don't work properly when set in the future |
ASTERISK-24443: CDR fields (dst, dcontext) empty in transfer call started from Macro |
ASTERISK-24444: PBX: Crash when generating extension for pattern matching hint |
ASTERISK-24445: [patch] queuerules option penaltychange ignored with ringall strategy |
ASTERISK-24446: logger.conf |
ASTERISK-24447: Bridge DTMF hooks: Audio doesn't pass when waiting for more matching digits. |
ASTERISK-24448: Crash on multi-leg call with directmedia=yes using pjsip |
ASTERISK-24449: Reinvite for T.38 UDPTL fails if SRTP is enabled |
ASTERISK-24450: Random Segmentation fault crash due to static realtime |
ASTERISK-24451: chan_iax2: reference leak in sched_delay_remove |
ASTERISK-24453: manager: acl_change_sub leaks |
ASTERISK-24454: app_queue: ao2_iterator not destroyed, causing leak |
ASTERISK-24455: func_cdr: CDR_PROP leaks payload |
ASTERISK-24456: SIP deadlock in transfer scenario between Asterisk Servers |
ASTERISK-24457: res_fax: fax gateway frames leak |
ASTERISK-24458: chan_phone fails to build on big endian systems |
ASTERISK-24459: bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible |
ASTERISK-24460: Make clean break build |
ASTERISK-24461: Parked calls end up in default parking lot |
ASTERISK-24462: res_pjsip: Stale qualify statistics after disablementation |
ASTERISK-24463: Voicemail email address corrupt or not sent when message is in the process of being recorded during reload |
ASTERISK-24464: [patch]ICE Candidates Gathering causes abort in pjproject code due to too small network interface array size |
ASTERISK-24465: audiohooks list leaks reference to formats |
ASTERISK-24466: app_queue: fix a couple leaks to struct call_queue |
ASTERISK-24467: STUN Request is not completing from ice_reset_session |
ASTERISK-24468: Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly) national symbols |
ASTERISK-24469: Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through |
ASTERISK-24470: No Transfer Event in Queue_log after unattended transfer |
ASTERISK-24471: Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2 |
ASTERISK-24472: Asterisk Crash in OpenSSL when calling over WSS from JSSIP |
ASTERISK-24473: Crash if a PRI DAHDI span is removed in the middle of a call |
ASTERISK-24474: sip_to_pjsip.py lacks documentation and does not function |
ASTERISK-24475: Wiki: Reference Count Debugging needs updating |
ASTERISK-24476: main/app.c / app_voicemail: ast_writestream leaks |
ASTERISK-24477: segfault in ast_translate at translate.c during calls with codec_siren7 |
ASTERISK-24478: IAX hits maxcallnumber limit of 2048 with only one concurrent channel |
ASTERISK-24479: Enable REF_DEBUG for module references |
ASTERISK-24480: res_http_websockets: Module reference decrease below zero |
ASTERISK-24481: Asterisk PJSIP 16-18 second delay in reply to INVITE |
ASTERISK-24482: func_talkdetect: Fix stasis message leak in audiohook callback |
ASTERISK-24483: res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id == -1 |
ASTERISK-24484: Update documentation for statsd module - usage requirements unclear |
ASTERISK-24485: res_pjsip cannot be unloaded or shutdown |
ASTERISK-24486: PJSIP contact rewriting fails to account for TCP timeout |
ASTERISK-24487: configuration: sections should be loadable as template even when not marked |
ASTERISK-24488: Wrong remote identity and target in dialog package XML in NOTIFY |
ASTERISK-24489: Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1 |
ASTERISK-24490: Security Vulnerability: CONFBRIDGE function's record_command option allows arbitrary parameters to be passed to MixMonitor, allowing remote execution of commands |
ASTERISK-24491: Memory leak in res_hep |
ASTERISK-24492: main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref |
ASTERISK-24493: Seg fault in pjsip_inv_send_msg from /usr/local/lib/libpjsip-ua.so.2 |
ASTERISK-24494: Setting timezone option in cdr_mysql to UTC has no effect |
ASTERISK-24495: Re-invite not changing RTP stream |
ASTERISK-24496: Asterisk random crash: when /var/lib/asterisk/sounds/ is mapped to windows share |
ASTERISK-24497: Hints change to Idle from InUse after sip reload. |
ASTERISK-24498: Segmentation fault in res_hep_rtcp on attended transfer |
ASTERISK-24499: Need more explicit debug when PJSIP dialstring is invalid |
ASTERISK-24500: Regression introduced in chan_mgcp by SVN revision r227276 |
ASTERISK-24501: ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd |
ASTERISK-24502: Build fails when dev-mode, dont optimize and coverage are enabled |
ASTERISK-24503: JSON assertion in stasis_channels - channel_blob_dtor |
ASTERISK-24504: chan_console: Fix reference leaks to pvt |
ASTERISK-24505: manager: http connections leak references |
ASTERISK-24507: MixMonitor transmit audio file corrupt after blind transfer |
ASTERISK-24508: pjsip - REFER request from SNOM is rejected with "400 bad request" - DEBUG shows "Received a REFER without a parseable Refer-To" |
ASTERISK-24509: Wrong request Line with my ISP Sip Trunk |
ASTERISK-24510: chan_sip: Add missing braces |
ASTERISK-24511: Not possible to Sign Digium License because no field for email |
ASTERISK-24512: asterisk -rx "module reload" + flag DEBUG_THREADS causes a deadlock |
ASTERISK-24513: Local channel apparently leaked in off-nominal DTMF attended transfer |
ASTERISK-24514: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard |
ASTERISK-24515: Unconditional use of fopencookie() / funopen() is non-portable |
ASTERISK-24516: [patch]Asterisk segfaults when playing back voicemail under high concurrency with an IMAP backend |
ASTERISK-24517: TLS support for Solaris, Ming and non-glibc Linux systems |
ASTERISK-24518: Avaya Asterisk sip trunk DTMF mis negotiated |
ASTERISK-24519: res_pjsip_phoneprov_provider doesn't get config from realtime. |
ASTERISK-24520: [patch]res_pjsip_phoneprov_provider.so has no realtime backend support |
ASTERISK-24521: [patch] Segfault due to null pointer in ast_bridged_channel |
ASTERISK-24522: ConfBridge: delay occurs between kicking all endmarked users when last marked user leaves |
ASTERISK-24523: Wiki Documentation - Review, Organize, Update the Configuration/Dialplan section |
ASTERISK-24524: Wiki Documentation - Review, Organize, Update the Configuration/Features wiki section |
ASTERISK-24525: Wiki Documentation - Review, Organize, Update the Configuration/Applications wiki section |
ASTERISK-24526: Exec dial from AGI script does not hang up caller after called party hangs up |
ASTERISK-24527: Give StopMixMonitor the ability to Stop MixMonitor on ANY channel. |
ASTERISK-24528: res_pjsip_refer: Sending INVITE with Replaces in-dialog with invalid target causes crash |
ASTERISK-24529: Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used |
ASTERISK-24530: [patch] app_record stripping 1/4 second from recordings |
ASTERISK-24531: res_pjsip_acl: ACLs not applied on initial module load |
ASTERISK-24532: res_phoneprov should automatically pull new phones from providers |
ASTERISK-24533: 2 threads created per chan_sip entry |
ASTERISK-24534: [patch]Register DB() as escalating to prevent users from writing to astdb |
ASTERISK-24535: stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix |
ASTERISK-24536: AMI redirect with PJSIP fails to move extra channel |
ASTERISK-24537: Stasis: StasisStart/StasisEnd events are not reliably transmitted during transfers |
ASTERISK-24538: SRTP p->tag corruption |
ASTERISK-24539: Compile fails on OSX because of sem_timedwait in bridge_channel.c |
ASTERISK-24540: chan_sip: inefficient database lookups when looking up a peer causes system slowdown with realtime |
ASTERISK-24541: Code that adds Required: timers to a 200 OK response will not work |
ASTERISK-24542: [patch]Failure showing codecs via 'core show channeltype <tech>' |
ASTERISK-24543: Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs |
ASTERISK-24544: Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll |
ASTERISK-24545: Monitor/Mixmonitor sip video calls |
ASTERISK-24546: Use Asterisk console: Conditional jump or move depends on uninitialised value |
ASTERISK-24547: CLONE - Make clean break build |
ASTERISK-24548: duration of the call in error callback. Can't send 10 type frames with SIP write |
ASTERISK-24549: on 'iax2 reload' all peers are announced as unreachable via AMI |
ASTERISK-24550: res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake |
ASTERISK-24551: Many diffeent valgrind issues in Asterisk |
ASTERISK-24552: ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes |
ASTERISK-24553: ARI/AMI: Include language in standard channel snapshot output |
ASTERISK-24554: AMI/ARI: Generate events on connected line changes |
ASTERISK-24555: Memory leak with T.38 fax and SLINEAR format |
ASTERISK-24556: Asterisk 13 core dumps when calling from pjsip extension to another pjsip extension |
ASTERISK-24557: WebRTC call returns error "Failed to get local SDP" |
ASTERISK-24558: Asterisk restarts all services |
ASTERISK-24559: app_voicemail notify_new_message() forked process crashes on OS X |
ASTERISK-24560: Creating a named ARI bridge twice causes a crash |
ASTERISK-24561: Core dump on res_fax_spandsp |
ASTERISK-24562: app_voicemail: Cannot set fromstring on a per-mailbox basis |
ASTERISK-24563: Direct Media calls within private network sometimes get one way audio |
ASTERISK-24564: Asterisk stops responding for a few minutes |
ASTERISK-24565: kqueue not working correctly on OSX |
ASTERISK-24566: Uninit buf in WS write |
ASTERISK-24567: Error loading module 'res_monitor.so': /usr/lib64/asterisk/modules/res_monitor.so: undefined symbol: __ast_beep_stop |
ASTERISK-24568: res_config_odbc fails on MS SQL Server |
ASTERISK-24569: user=phone is not added to From, Contact and Diversion header |
ASTERISK-24571: Deadlock when asterisk is loading and CLI command "sip reload" is executed |
ASTERISK-24572: [patch]App_meetme is loaded without its defaults when the configuration file is missing |
ASTERISK-24573: [patch]Out of sync conversation recording when divided in multiple recordings |
ASTERISK-24574: Bouncy transfer tests |
ASTERISK-24575: [patch]Make capath work for res_pjsip |
ASTERISK-24576: SLA calls causing segfault in Channel.c |
ASTERISK-24577: Speed up loopback switches by avoiding unneeded lookups |
ASTERISK-24578: TestSuite: Write Attended Transfers from non-Stasis to a Stasis bridge |
ASTERISK-24579: TestSuite: Write Attended Transfers from Stasis Bridge to non-Stasis |
ASTERISK-24580: TestSuite: Write Attended Transfers from Stasis Bridge to Stasis |
ASTERISK-24581: TestSuite: Write Blind Transfer tests for Stasis application interaction |
ASTERISK-24582: Realtime with multiple DB destinations using priorities results in duplicate queries |
ASTERISK-24583: TestSuite: Write External Bridging test for Stasis application - Stasis bridge with dialplan application |
ASTERISK-24584: RTP QOS Channel variables not being set correctly |
ASTERISK-24585: One way speech when performing attended Xfer issue appears to be SIP invite related when switching back from native_rtp bridge |
ASTERISK-24586: TestSuite: Write channel subscription tests for ARI |
ASTERISK-24587: Invalid free in valgrind output |
ASTERISK-24588: res_calendar does not process CalDAV from Owncloud [fix included] |
ASTERISK-24589: Inefficient Translation from g722 to silk16 via slin |
ASTERISK-24590: Segmentation fault in get_multiple_fields_as_var_list |
ASTERISK-24591: Stasis() side of an ARI originated channel cannot be Redirected |
ASTERISK-24592: Random crashes with preceding ERROR "Failed to encode JSON object" |
ASTERISK-24593: Asterisk Hangs, All DAHDI Channels Off-Hook Or Ringing |
ASTERISK-24594: SIP Signal 183 Session Progress doesn't update Channel status to |
ASTERISK-24595: chan_sip.c Deadlock on SMP |
ASTERISK-24596: Unclear how to use Park application with res_parking 'parkeddynamic' enabled. Documentation? |
ASTERISK-24597: install_prereq script does not install sqlite 3 on Centos6 |
ASTERISK-24598: When running ./contrib/scripts/install_prereq install-unpackaged pjproject is installed in wrong place |
ASTERISK-24599: Asterisk SIP Freezes |
ASTERISK-24600: Stuck IAX channels, Asterisk stops responding to most traffic, potential deadlock |
ASTERISK-24601: [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY event: dialog XML body |
ASTERISK-24602: Unable to call WebRTC client via wss on chan_pjsip |
ASTERISK-24603: Channels are stucked in Ring State on Dial Application |
ASTERISK-24604: res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core |
ASTERISK-24605: res_parking option parkeddynamic does not work with the core Features 'parkcall' (DTMF initiated parking) |
ASTERISK-24606: Memory corruption issue which started asterisk 1.6.1.6 after every 20,30 mins |
ASTERISK-24607: res_pjsip_session: re-INVITE with declined media streams results in 488 |
ASTERISK-24608: Asterisk consumes more memory (and CPU) gradually: memory leak. |
ASTERISK-24609: Features DTMF parking to an existing custom parkinglot with no 'parkext' defined results in silent failure |
ASTERISK-24610: TestSuite: Write External Bridging test for Stasis bridge (one channel) interactions |
ASTERISK-24611: TestSuite: Write External Bridging test for Stasis (two channel) interactions |
ASTERISK-24612: res_pjsip: No information if a required sorcery wizard is not loaded |
ASTERISK-24613: Wrong NewExten Manager Event sent directly after features.conf Blind Transfer |
ASTERISK-24614: Deadlock when DEBUG_THREADS compiler flag enabled |
ASTERISK-24615: When Multiple Transports Exist in pjsip.conf, Incorrect External Addresses is Used in SIP Packets When Responding to INVITE |
ASTERISK-24616: Crash in res_format_attr_h264 due to invalid string copy |
ASTERISK-24617: Asterisk 11.14.1 core dump |
ASTERISK-24618: Asterisk goes down with segfault message |
ASTERISK-24619: [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int |
ASTERISK-24620: ANSWEREDTIME channel variable read via AGI returns NULL result on an answered channel |
ASTERISK-24621: chan_sip: Crash caused by invalid reference to object in __sip_autodestruct |
ASTERISK-24622: chan_sip: Crash when disposing of dialog in scheduled callback __sip_autodestruct |
ASTERISK-24623: Hard coded PJDIR in res/pjproject/build.mak |
ASTERISK-24624: Transfer to invalid extension results in hung channel. |
ASTERISK-24625: res_pjsip_outbound_registration: 401 spam when invalid credentials provided |
ASTERISK-24626: Voicemail passwords not being stored in ARA |
ASTERISK-24627: Can't take more than 11 voicemails simultaneously when using realtime |
ASTERISK-24628: [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment) |
ASTERISK-24629: Asterisk crashing randomly, appears related to res_hep_rtcp |
ASTERISK-24630: When defining 'parkext' for a custom lot, 'parkext_exclusive' is required to have the parking extension park at the custom lot |
ASTERISK-24631: Incorrect description of option "context" in queues.conf.sample |
ASTERISK-24632: install_prereq script installs pjproject without IPv6 support |
ASTERISK-24633: Asterisk crashing (awaiting backtrace) |
ASTERISK-24634: cdr_adaptive_odbc fails to enter data in db |
ASTERISK-24635: PJSIP outbound PUBLISH crashes when no response is ever received |
ASTERISK-24636: TestSuite: Inbound nominal registration test bounce due to multiple registration refresh test events |
ASTERISK-24637: Channel re-enters Stasis() when it should not |
ASTERISK-24638: Crash after destroying a stasis bridge |
ASTERISK-24639: Crash with PJSIP on SIP to SIP over WebSockets call (WebRTC, SIPML5) |
ASTERISK-24640: Registration pending stays forever after sip reload |
ASTERISK-24641: Deadlock in Trunk |
ASTERISK-24642: 13 branch broken, can't make SIP calls |
ASTERISK-24643: res_pjsip: Add user=phone option |
ASTERISK-24644: res_pjsip_keepalive: Add keepalive module for connection-oriented transports. |
ASTERISK-24645: Cdr record Not saving after upgrade Asterisk 1.6.24 to Asterisk 11.15.0 |
ASTERISK-24646: PJSIP changeset 4899 breaks TLS |
ASTERISK-24647: Random Segfaults |
ASTERISK-24648: Execiftime() not support appiffalse |
ASTERISK-24649: Pushing of channel into bridge fails; Stasis fails to get app name |
ASTERISK-24650: [patch] Fix access check to structures in rtp_engine.c |
ASTERISK-24651: [patch] Fix race condition in DTLS |
ASTERISK-24652: Confbridge interaction with CDR |
ASTERISK-24653: applicationmap not recognized |
ASTERISK-24654: Cant Register when extra data is added to Registers URI using PJSIP |
ASTERISK-24655: res_pjsip_outbound_publish: Hang on shutdown while attempting to publish |
ASTERISK-24656: AMI: Protocol errors |
ASTERISK-24657: directmedia=no does not work in Asterisk 13.1.0 |
ASTERISK-24658: SILK Codec binary for Asterisk 13 |
ASTERISK-24659: Execiftime() add false argument |
ASTERISK-24660: Crash during AMI Originate with DEBUG_THREADS enabled |
ASTERISK-24661: Wiki Documentation - Edits and updates to document Asterisk's audio and video capabilities (formats, codecs) |
ASTERISK-24662: [patch] column and row headers for Signed Linear format variants in output of 'core show translation' are ambiguous |
ASTERISK-24663: [patch] Unnamed semaphore autoconf check fails on cross compilation |
ASTERISK-24664: PJSIP Error processing packet on SIP requests with non-ascii characters in INVITE or REGISTER |
ASTERISK-24665: Configure check required for pjsip_get_dest_info() |
ASTERISK-24666: Security Vulnerability: RTP not closed after sip call using unsupported codec |
ASTERISK-24667: IAX hits maxcallnumber limit of 2048 after many calls and don't recover after hangup |
ASTERISK-24668: app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Subscriber absent) |
ASTERISK-24669: crash - memory corruption, or invalid channel snapshot in a stasis message |
ASTERISK-24670: Asterisk 13.1.0 with webRTC missing ice-ufrag and ice-pwd compiling |
ASTERISK-24671: Missing docs for the CDR AMI Event |
ASTERISK-24672: [PATCH] Memory leak in func_curl CURLOPT |
ASTERISK-24673: outgoing sip registers cannot be removed or modified without doing restart (or doing module unload chan_sip.so) |
ASTERISK-24674: CDR backends not documented very well |
ASTERISK-24675: IAX2 sees peer as not connected but replies to POKE |
ASTERISK-24676: Security Vulnerability: URL request injection in libCURL (CVE-2014-8150) |
ASTERISK-24677: ARI GET variable on channel provides unhelpful response on non-existent variable |
ASTERISK-24678: [PATCH] Added atxfer* settings to features.conf.sample |
ASTERISK-24679: Wrong IAX2 context chosen due to netmask |
ASTERISK-24680: [patch] - Added FAXOPT(gateway)=t38 option |
ASTERISK-24681: Asterisk restarts during attended transfer if on-hold peer is disconnected before the ringing tones are heard |
ASTERISK-24682: app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value |
ASTERISK-24683: Crash in PBX ast_hashtab_lookup_internal during core restart now |
ASTERISK-24684: crash during sip reload |
ASTERISK-24685: "pjsip show version" CLI command |
ASTERISK-24686: The problem with the correct sample exten in realtime mode |
ASTERISK-24687: Asterisk behind NAT sets wrong Contact Header |
ASTERISK-24688: res_pjsip: Add a "run once" ability to supplements |
ASTERISK-24689: Segfault on hangup after outgoing PRI-Euroisdn call |
ASTERISK-24690: crash during sip reload |
ASTERISK-24691: Asterisk tries to transcode between g722 & h264 |
ASTERISK-24692: PJSIP: Provisional 181 response during a call forward is given an SDP, which cancels ringing on devices without Asterisk providing an inband indication |
ASTERISK-24693: Investigate and fix memory leaks in Asterisk |
ASTERISK-24694: CLONE - Cant Register when extra data is added to Registers URI using PJSIP |
ASTERISK-24695: Unable to get 'B' function of MixMonitor to work with ChanSpy |
ASTERISK-24696: testsuite: sip_attended_transfer test is bouncing |
ASTERISK-24697: MixMonitor doesn't work properly with the option: b |
ASTERISK-24698: Asterisk 13.1.0 PJSIP over TCP gives PJSIP_ETPNOTSUITABLE |
ASTERISK-24699: VoiceMailMain hangs up with no reason when trying to play first message |
ASTERISK-24700: CRASH: NULL channel is being passed to ast_bridge_transfer_attended() |
ASTERISK-24701: Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information |
ASTERISK-24702: TestSuite: registration inbound nominal multiple test triggers scheduler assertion in contact_expiration_observer_deleted |
ASTERISK-24703: ARI: Add the ability to "transfer" (redirect) a channel |
ASTERISK-24704: Inconsistency in Voicemail store on phone notification |
ASTERISK-24705: No sound when using WebRTC in some calls |
ASTERISK-24706: [patch]add auto-dtmf mode for pjsip |
ASTERISK-24707: Double free corruprion in PJSIP |
ASTERISK-24708: Unable to connect to the asterisk CLI and after a while asterisk creates a core dump |
ASTERISK-24709: [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event |
ASTERISK-24710: Call forwarding to misc destination works only with first trunk if CID number is stated in dialing pattern |
ASTERISK-24711: DTLS handshake broken with latest OpenSSL versions |
ASTERISK-24712: xmpp: starttls problem causes connection spew |
ASTERISK-24713: CALLERID(ani2) needs to be read/write |
ASTERISK-24714: Asterisk crashes when other device does not accept min-se and sends 422 response code |
ASTERISK-24715: chan_sip: stale nonce causes failure |
ASTERISK-24716: Improve pjsip log messages for presence subscription failure |
ASTERISK-24717: ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} |
ASTERISK-24718: [patch]Add inital support of "sanitize" to configure |
ASTERISK-24719: ConfBridge recording channels get stuck when recording started/stopped more than once |
ASTERISK-24720: Unable to change voicemail password using realtime engine |
ASTERISK-24721: manager: ModuleLoad action incorrectly reports 'module not found' during a Reload operation |
ASTERISK-24722: Call terminates without executing 'h' extension |
ASTERISK-24723: confbridge: CLI command 'confbridge list XXXX' no longer displays user menus |
ASTERISK-24724: 'httpstatus' Web Page Produces Incomplete HTML |
ASTERISK-24725: WebSockets uses first loaded SIP module(chan_sip, pjsip) as SIP provider for WebRTC |
ASTERISK-24726: Cleaned ast_channel in file.c waitstream_core |
ASTERISK-24727: PJSIP: Crash experienced during multi-Asterisk transfer scenario. |
ASTERISK-24728: tcptls: Bad file descriptor error when reloading chan_sip |
ASTERISK-24729: Outbound registration not occuring on new registrations after reload. |
ASTERISK-24730: [patch] Add blank line between headers and output for Command action response |
ASTERISK-24731: res_pjsip_session cannot be unloaded |
ASTERISK-24732: [patch]CLI "core show channel" gets "failed to extend from %d to %d" message for large native formats list. |
ASTERISK-24733: CLI "module reload cdr" does not handle an unchanged config file very well. |
ASTERISK-24734: 'config list' output not so useful for humans - pjsip.conf entries list association with a unique ID instead of a module/component |
ASTERISK-24735: [patch] - Video Media support broken for (WebRTC endpoints) |
ASTERISK-24736: Memory Leak Fixes |
ASTERISK-24737: When agent not logged in, agent status shows unavailable, queue status shows agent invalid |
ASTERISK-24738: Disconnect option in features.conf not functioning in Asterisk 13.1.0 |
ASTERISK-24739: [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules |
ASTERISK-24740: [patch]Segmentation fault on aoc-e event |
ASTERISK-24741: dtls_handler causes Asterisk to crash |
ASTERISK-24742: [patch] Fix ast_odbc_find_table function in res_odbc |
ASTERISK-24743: REG implementation of Asterisk |
ASTERISK-24744: Swedish Core Voice prompts |
ASTERISK-24745: [patch]Add no_answer to ARI hangup causes |
ASTERISK-24746: asterisk not installing with libjansson dev package |
ASTERISK-24747: Call is not terminated in time due to lack of RTP nor it is shutdown if peer sends BYE |
ASTERISK-24748: res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur |
ASTERISK-24749: ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when played to bridge |
ASTERISK-24750: stun is in use although all conf files have icesupport=no |
ASTERISK-24751: Integer values in json payload to ARI cause asterisk to crash |
ASTERISK-24752: Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown |
ASTERISK-24753: warning happens when compiling with don't optimize |
ASTERISK-24754: Asterisk 13 crashes every few minutes and does not generate core |
ASTERISK-24755: Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge |
ASTERISK-24756: ConfBridge sound_muted does not work from CLI or AMI |
ASTERISK-24757: INTERNAL_OBJ: user_data is NULL and segfaults |
ASTERISK-24758: Lengthy CDR chain crashes during object destruction |
ASTERISK-24759: Reference of deleted ao2 object during shutdown of res_pjsip_pubsub |
ASTERISK-24760: Invalid pointer in ast_context_destroy during shutdown |
ASTERISK-24761: Asterisk channel locking up with high call traffic |
ASTERISK-24762: not able to make outbound call from Asterisk/1.8.13.1 |
ASTERISK-24763: CalDav calendar fails to process day long event when time is set to 00:00 |
ASTERISK-24764: AMI event notification of linkedid propagation |
ASTERISK-24765: retrans_pkt: Retransmission timeout reached / Hanging up calls |
ASTERISK-24766: [patch] rtcp-fb fir support missing from chan_pjsip |
ASTERISK-24767: Regression - fromdomain port ingnored in some situations |
ASTERISK-24768: res_timing_pthread: file descriptor leak |
ASTERISK-24769: res_pjsip_sdp_rtp: Local ICE candidates leaked |
ASTERISK-24770: Astersk segfault when reload SNMP service |
ASTERISK-24771: ${CHANNEL(pjsip)} - segfault |
ASTERISK-24772: ODBC error in realtime sippeers when device unregisters under MariaDB |
ASTERISK-24773: Hung call in "sip show channels" not listed in "show channels" |
ASTERISK-24774: Segfault in ast_context_destroy with extensions.ael and extensions.conf |
ASTERISK-24775: Segfault in Asterisk 13.0.0 During Call |
ASTERISK-24776: The year in the CLI copyright notice is now wrong |
ASTERISK-24777: chan_sip and chan_pjsip cannot coexist when using websocket transport |
ASTERISK-24778: OPUS codec not working with chan_pjsip |
ASTERISK-24779: Passthrough OPUS codec not working with chan_pjsip |
ASTERISK-24780: [patch] - Buddies are always auto-registered when processing the roster |
ASTERISK-24781: PJSIP: Unnecessary 180 Ringing messages sent with undesireabe consequences. |
ASTERISK-24782: StasisEnd event not present for channel that was swapped out for another after completing attended transfer |
ASTERISK-24783: Build fails with gcc 5 |
ASTERISK-24784: Asterisk v13.2.0 + chan_sip + leaks pipes |
ASTERISK-24785: 'Expires' header missing from 200 OK on REGISTER |
ASTERISK-24786: [patch] - Asterisk terminates when playing a voicemail stored in LDAP |
ASTERISK-24787: [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime |
ASTERISK-24788: Can't get res_snmp.so to load |
ASTERISK-24789: res_snmp.so cannot load |
ASTERISK-24790: Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context |
ASTERISK-24791: Crash in ast_rtcp_write_report |
ASTERISK-24792: CLONE - app_transfer fails with PJSIP channels |
ASTERISK-24793: Asterisk spirals to high load when audio stream fed to MOH disappears |
ASTERISK-24794: Spandsp FAX Driver Q? |
ASTERISK-24795: AEL - When a call is dialed using a gosub to a different context using the switch/goto statement and the call is parked, recovering the call gives a wrong callerid |
ASTERISK-24796: Codecs and bucket schema's prevent module unload |
ASTERISK-24797: bridge_softmix: G.729 codec license held |
ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor |
ASTERISK-24799: [patch] make fails with undefined reference to SSLv3_client_method |
ASTERISK-24800: Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill |
ASTERISK-24801: ASAN: ast_el_read_char stack-buffer-overflow |
ASTERISK-24802: stasis: set a channel variable on websocket disconnect error |
ASTERISK-24803: (Un)PauseQueueMember AMI action writes event to queue_log even on failure |
ASTERISK-24804: ASAN heap-buffer-overflow c_setpat |
ASTERISK-24805: [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing |
ASTERISK-24806: Absent audio between WebRTC clients in local network |
ASTERISK-24807: Missing mandatory field Max-Forwards |
ASTERISK-24808: res_config_odbc: Improper escaping of backslashes occurs with MySQL |
ASTERISK-24809: core show translation - gives (hopefully) erroneous output |
ASTERISK-24810: Cannot sent DTMF tones via H323 trunk as H245 signal. |
ASTERISK-24811: asterisk-publication sorcery object does not use realtime |
ASTERISK-24812: ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding |
ASTERISK-24813: asterisk.c: #if statement in listener() confuses code folding editors |
ASTERISK-24814: asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers |
ASTERISK-24815: [patch] Enable TLS Dual-Certificates (ECC+RSA) |
ASTERISK-24816: No fax for asterisk 13 |
ASTERISK-24817: init_logger_chain: unreachable code block |
ASTERISK-24818: PJSIP in dialog OPTIONS method handling |
ASTERISK-24819: res_odbc: pre-connect -> Floating point exception (core dumped) |
ASTERISK-24820: Asterisk Cert 13-1 crashing |
ASTERISK-24821: Question about Warning message |
ASTERISK-24822: Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code |
ASTERISK-24823: how to apply patch on asterisk |
ASTERISK-24824: chan_sip: Asterisk fails to re-activate an inactive media session when an offer does not contain a=sendrecv |
ASTERISK-24825: Caller ID not recognized using Centrex/Distinctive dialing |
ASTERISK-24826: Cannot sent DTMF tones via H323 trunk as H245 signal. |
ASTERISK-24827: Missing documentation for chan_dahdi dial string ring cadences |
ASTERISK-24828: Fix Frame Leaks |
ASTERISK-24829: I need Asterisk AMI in java which supports Asterisk 13,i am in big trouble.Any JAR,compiled file or java project can solve my problem. |
ASTERISK-24830: res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT |
ASTERISK-24831: Upon receiving BYE a dialog is scheduled for destruction in 32000 ms and gets stuck, causing long calls |
ASTERISK-24832: [patch]DTLS-crashes within openssl |
ASTERISK-24833: [patch] audit of startup order reveals logger concerns |
ASTERISK-24834: DNS Overhaul: Implement the proposed core API - sync/async functions, resolver registration |
ASTERISK-24835: Early Media Not working with Chan SIP and Asterisk 13 |
ASTERISK-24836: DNS Overhaul: Write a Resolver Implementation |
ASTERISK-24837: chan_sip calls to Asterisk result in file descriptors growing exponentially while channels remain up |
ASTERISK-24838: chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling |
ASTERISK-24839: CDRs: LinkedID field in Attended Transfer not reporting as expected |
ASTERISK-24840: res_pjsip: conflicting endpoint identifiers |
ASTERISK-24841: ConfBridge: Strange sampling rates chosen when channels have multiple native formats |
ASTERISK-24842: fax to email |
ASTERISK-24843: BSD = "Berkeley Software Distribution" not "Berkeley Science Division" |
ASTERISK-24844: chan_iax2: name of channels can cause Bridge action to use the wrong channels when bridging is made using the channel name |
ASTERISK-24845: pjsip send notify not working with Cisco phone |
ASTERISK-24846: Cancel Request Broken in chan_pjsip when it's used on Trunk with TCP transport |
ASTERISK-24847: [security] [patch] tcptls: certificate CN NULL byte prefix bug |
ASTERISK-24848: INVITE not send by asterisk on chan_pjsip when endpoint has TCP Transport |
ASTERISK-24849: AMI bridge command fails - ast_bridge_add_channel() function needs to check that the channel is not only in a bridge but that the bridge will allow channels to be moved from it. |
ASTERISK-24850: chan_lcr: Deadlock caused by chan_lock interaction with channel container lock and individual channel lock |
ASTERISK-24851: Asterisk checks both joinempty and leavewhenempty settings before caller enters queue and if a match is found in either executes EXITEMPTY |
ASTERISK-24852: [patch] new res_pjsip module to identify endpoint for an incoming call with a trunk that has outbound registration. |
ASTERISK-24853: Documentation claims chan_sip outbound registrations support WS or WSS as valid transports (not true) |
ASTERISK-24854: Asterisk with PJSIP segfaults after enabling TLS (SSL) transport |
ASTERISK-24855: TLS NOTIFY with SIPS uses "sip:sips" in To header |
ASTERISK-24856: Need Help asterisk Openais Corosysnc installtion configuartuion |
ASTERISK-24857: [patch] "timing test", pjsip incoming/outgoing calls, voicemail prompts and recordings all fail when using the kqueue timer source on FreeBSD 10.x |
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec |
ASTERISK-24859: Update requested to Security Log File Format wiki page |
ASTERISK-24860: Update requested to Asterisk Queue Log File Format wiki page |
ASTERISK-24861: callerid number does not updates in the CDR |
ASTERISK-24862: [patch] Support in-dialog OPTIONS |
ASTERISK-24863: res_pjsip: No endpoint events raised via AMI when contacts cannot be reached/qualified |
ASTERISK-24864: app_confbridge: file playback blocks dtmf |
ASTERISK-24865: Asterisk segfaults when paging speakers and phone - hanging up all lines |
ASTERISK-24866: [patch]Separate QoS settings for trunk and other packets |
ASTERISK-24867: Docs for 'e' option in ResetCDR say to use CDR_PROP instead, CDR_PROP docs are unclear |
ASTERISK-24868: Bad audio with silk if connection is bridged over third server and g722 on one side |
ASTERISK-24869: Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel |
ASTERISK-24870: ARI: Subscriptions to bridges generally not super useful |
ASTERISK-24871: [patch] build_peer peer mailbox management bug |
ASTERISK-24872: [patch] AMI PJSIPShowEndpoint closes AMI connection on error |
ASTERISK-24873: Cancelling transmission of IAX2 packet loop wrap around |
ASTERISK-24874: Asterisk 11/13 Named ACL misconfiguration produces misleading errors - lacking commands to debug or troubleshoot |
ASTERISK-24875: Randomly get segfaults processing WEBRTC calls with app_konference |
ASTERISK-24876: Investigate reference leaks from tests/channels/local/local_optimize_away |
ASTERISK-24877: Subsequent MWI NOTIFY R-URI truncated |
ASTERISK-24878: Compilation fails due to Backtrace under OpenBSD |
ASTERISK-24879: [patch]Compilation fails due to 64bit time under OpenBSD |
ASTERISK-24880: [patch]Compilation under OpenBSD |
ASTERISK-24881: ast_register_atexit should only be used when absolutely needed |
ASTERISK-24882: chan_sip: Improve usage of REF_DEBUG |
ASTERISK-24883: HTTP max content length - repercussions on ARI service |
ASTERISK-24884: Crash in ast_bridge_transfer_attended |
ASTERISK-24885: Frequent Asterisk 13.2 core crash |
ASTERISK-24886: App_followme "f" option - basic follow_me |
ASTERISK-24887: [patch]tags in a=crypto lines do not accept 2 or more digits |
ASTERISK-24888: Revision 422070 and 425503 are not merged to Asterisk 11 branch |
ASTERISK-24889: Possible bug: PJSIP: stateless behavior when transport=tcp |
ASTERISK-24890: res_pjsip_acl: patch proposal - endpoint specific ACL |
ASTERISK-24891: [USAN] Int overflow in strings.h |
ASTERISK-24892: Super Awesome Company sound prompts |
ASTERISK-24893: PJSIP unhandled exception PJLIB/No memory |
ASTERISK-24894: [patch] iax2_poke_noanswer expiration timer too short |
ASTERISK-24895: After hangup on the side of the ISDN network no HangupRequest event comes for the dahdi channel. |
ASTERISK-24896: [patch] Using force black background leads to colours not being reset |
ASTERISK-24897: Confbridge dtmf_passthrough is not respected when using menus |
ASTERISK-24898: Confbridge ignores default_user/bridge when using CONFBRIDGE function |
ASTERISK-24899: Parking fall-through behavior different in 13 |
ASTERISK-24900: Manager event ParkedCallSwap is not documented |
ASTERISK-24901: OOH323 get UUI information |
ASTERISK-24902: spontaneous asterisk processes fall |
ASTERISK-24903: Call just hangup after tried to redirect from parkinglot. |
ASTERISK-24904: cam-vm-call1 hung and was restarted manually [3/23/2015 5:33pm ET] |
ASTERISK-24905: wrong mutex macros in include/asterisk/lock.h |
ASTERISK-24906: [patch] Asterisk doesn't compile under Cygwin |
ASTERISK-24907: res_pjsip_outbound_registration: crash during unload if registration attempts are still occuring |
ASTERISK-24908: UUI sent from Avaya to Asterisk over ISDN |
ASTERISK-24909: Add clustering of the AstDB using PJSIP |
ASTERISK-24910: "timer=no" and "timer=required" settings in pjsip.conf fail |
ASTERISK-24911: No Voice with Firefox-37 as DTLS handshake is not completing |
ASTERISK-24912: pjsip segmentation fault in pjmedia_sdp_attr_clone(../src/pjmedia/sdp.c:134) |
ASTERISK-24913: CLONE - missing Contact header in 200 OK to INVITE |
ASTERISK-24914: Division by zero in file.c when playback of voicemail with video as h264 |
ASTERISK-24915: [patch]Missing Contact: header in 200 OK |
ASTERISK-24916: Increasing memory usage when multiple reinvite during call |
ASTERISK-24917: [patch] clang compilation warnings |
ASTERISK-24918: pjsip: add CLI options to display global and system configuration |
ASTERISK-24919: res_pjsip_config_wizard: Ability to write contents to file |
ASTERISK-24920: Asterisk handles duplicate SIP requests as if they were each a new request |
ASTERISK-24921: [patch]RAII reentrancy issue when compiled with clang |
ASTERISK-24922: ARI: Add the ability to intercept hold and raise an event |
ASTERISK-24923: Enabling DONT_OPTIMIZE can cause link errors to be lost. |
ASTERISK-24924: [patch]res_pjsip_exten_state - logging swapped argument |
ASTERISK-24925: Crash within pjprojects(libpjnath) pj_stun_session_on_rx_pkt |
ASTERISK-24926: CLONE - Asterisk listening/communicating on undefined IP |
ASTERISK-24927: app_voicemail (IMAP support) function save_to_folder: creates wrong folder |
ASTERISK-24928: [patch]t38_udptl_maxdatagram in pjsip.conf not honored |
ASTERISK-24929: WebRTC Client No Audio after 60 seconds on Windows Only |
ASTERISK-24930: SayNumber() gender option get error and hangup channel. |
ASTERISK-24931: dns: Add support for SRV records. |
ASTERISK-24932: Asterisk 13.x does not build with GCC 5.0 |
ASTERISK-24933: T38 fails negotiation |
ASTERISK-24934: [patch]Asterisk manager output does not escape control characters |
ASTERISK-24935: res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator. |
ASTERISK-24936: New Feature: AO2 weakproxy objects |
ASTERISK-24937: [patch]res_pjsip_messaging: Messages may be sent out of order |
ASTERISK-24938: ARI Snoop Channel results in excessive escalating CPU usage |
ASTERISK-24939: [patch]IAX make calltoken expiration time configurable |
ASTERISK-24940: cdr |
ASTERISK-24941: Asterisk 13: ABI compatibility issue in res_pjsip_session breaks external modules |
ASTERISK-24942: Voicemail API: message is deleted when destination mailbox is at maxmsg |
ASTERISK-24943: Wrong ANSWEREDTIME from Dial app when the channel was answered earlier |
ASTERISK-24944: main/audiohook.c change prevents G722 call recording |
ASTERISK-24945: Apply PJSIP global default values when no global configuration is present |
ASTERISK-24946: asterisk 13.2 and pjsip calls hangup twice |
ASTERISK-24947: res_pjsip: Add a PJSIP resolver using core DNS |
ASTERISK-24948: Review Board 4457 isn't merged to Asterisk 11 branch |
ASTERISK-24949: res_pjsip_outbound_registration: Backport line functionality |
ASTERISK-24950: ChanSpy B option doesn't work perfectly |
ASTERISK-24951: Asterisk crash off in segmenation fault |
ASTERISK-24952: global timerb is disregarded by chan_sip |
ASTERISK-24953: segmentation fault after changing PJSIP_MAX_PKT_LEN |
ASTERISK-24954: Git migration: Asterisk version numbers are incompatible with the Test Suite |
ASTERISK-24955: res_fax: v.27ter support baud rate of 2400, which is disallowed in res_fax's check_modem_rate |
ASTERISK-24957: match_auth_username=yes doesn't work |
ASTERISK-24958: Forwarding loop detection inhibits certain desirable scenarios |
ASTERISK-24959: [patch]CLI command cdr show pgsql status |
ASTERISK-24960: Build System: Create MOD_ADD_SOURCE macro for module Makefiles |
ASTERISK-24961: segmenation fault in dtls_perform_handshake |
ASTERISK-24962: Asterisk does not compile with last pjproject |
ASTERISK-24963: ASAN: heap-use-after-free with PJSIP and WSS |
ASTERISK-24964: memory overlap in pj_gethostip |
ASTERISK-24965: cel_pgsql - log_error string references CDR instead of CEL |
ASTERISK-24966: Document limitation of MessageSend regarding AOR destination |
ASTERISK-24967: Problem support schema for pgsql on CEL |
ASTERISK-24968: Deadlock |
ASTERISK-24969: Named ACL's do not handle config errors. |
ASTERISK-24970: Crash in res_pjsip_pubsub handling of failed notify |
ASTERISK-24971: Asterisk crashes on voicemail |
ASTERISK-24972: Transport Layer Security (TLS) Protocol BEAST Vulnerability - Investigate vulnerability of HTTP server |
ASTERISK-24973: Wacky Send: Pack Lost ( %) Jitter |
ASTERISK-24974: Astobj2: Allow reference debugging to be enabled/disabled by config. |
ASTERISK-24975: Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail |
ASTERISK-24976: cdr_odbc not include new columns added on 1.8 |
ASTERISK-24977: Contacts that don't use qualify are being marked as unavailable |
ASTERISK-24978: testsuite: Refactor multiple occurrences of AMISendTest in tests/channels/pjsip |
ASTERISK-24979: Webrtc client audio output is consistently skipping or missing non-continuous audio |
ASTERISK-24980: cdr_adaptive_odbc: refactor lines to concatenate of columns name |
ASTERISK-24981: PJSIP: In-dialog NOTIFYs sent to private NAT IP address |
ASTERISK-24982: res_pjsip_mwi: Unsolicited MWI NOTIFY only sent on mailbox changes |
ASTERISK-24983: IAX deadlock between hangup and scheduled actions (ex. largrq) |
ASTERISK-24984: testsuite: run-local still relies on outside components |
ASTERISK-24985: Masquerade test is failing in 13+ |
ASTERISK-24986: keepalive INFO packages ignored by asterisk |
ASTERISK-24987: res_pjsip: Resolver tests do not have resolver as a dependency |
ASTERISK-24988: func_talkdetect: Test is bouncing sporadically |
ASTERISK-24989: Deadlock between chan_sip, channels, and pbx_realtime |
ASTERISK-24990: asterisk Crash Bad magic number |
ASTERISK-24991: Check for ao2_alloc failure in __ast_channel_internal_alloc |
ASTERISK-24992: [patch] cdr_adaptive_odbc: Check the column in table is nulleable |
ASTERISK-24993: No forum signup confirmation email received |
ASTERISK-24994: dns: Query set unit tests are failing due to race condition |
ASTERISK-24995: Testsuite Test 'stasisstatus' Fails During Build |
ASTERISK-24996: chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf |
ASTERISK-24997: Astobj2: Some callers of __adjust_lock do not pre-check the object |
ASTERISK-24998: res_corosync: res_corosync tries to load even if res_corosync.conf is missing |
ASTERISK-24999: PJSIP crashes with malformed contact line |